[FFmpeg-devel] [PATCH 3/5] mp3dec: parse initial silence padding information from lame tag

[Nicolas George]

Le septidi 17 messidor, an CCXX, Michael Niedermayer a écrit :
> And pass the information on to the lavf core.
> 
> Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> ---
>  libavformat/mp3dec.c |   28 ++++++++++++++++++++++++++++
>  1 file changed, 28 insertions(+)
> 
> diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
> index 61405f7..bc62694 100644
> --- a/libavformat/mp3dec.c
> +++ b/libavformat/mp3dec.c
> @@ -31,6 +31,8 @@
>  
>  typedef struct {
>      int64_t filesize;
> +    int start_pad;
> +    int end_pad;

They look to be used only locally (and, for end_pad, only in debug output).
Do you have a reason to put them in the context?

>  } MP3Context;
>  
>  /* mp3 read */
> @@ -82,6 +84,7 @@ static int mp3_read_probe(AVProbeData *p)
>   */
>  static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
>  {
> +    MP3Context *mp3 = s->priv_data;
>      uint32_t v, spf;
>      unsigned frames = 0; /* Total number of frames in file */
>      unsigned size = 0; /* Total number of bytes in the stream */
> @@ -107,6 +110,20 @@ static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
>              frames = avio_rb32(s->pb);
>          if(v & 0x2)
>              size = avio_rb32(s->pb);
> +        if(v & 4)
> +            avio_skip(s->pb, 100);
> +        if(v & 8)
> +            avio_skip(s->pb, 4);
> +
> +        v = avio_rb32(s->pb);
> +        if(v == MKBETAG('L', 'A', 'M', 'E')) {
> +            avio_skip(s->pb, 21-4);
> +            v= avio_rb24(s->pb);
> +            mp3->start_pad = v>>12;
> +            mp3->  end_pad = v&4095;
> +            st->skip_samples = mp3->start_pad + 528 + 1;
> +            av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3->  end_pad);
> +        }
>      }
>  
>      /* Check for VBRI tag (always 32 bytes after end of mpegaudio header) */
> @@ -206,6 +223,16 @@ static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
>      return ret;
>  }
>  
> +static int read_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags)
> +{
> +    MP3Context *mp3 = s->priv_data;
> +    AVStream *st = s->streams[stream_index];
> +
> +    st->skip_samples = (mp3->start_pad + 528 + 1) * (timestamp <= 0);

st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0

would look more readable to my eyes.

> +
> +    return -1;
> +}
> +
>  AVInputFormat ff_mp3_demuxer = {
>      .name           = "mp3",
>      .long_name      = NULL_IF_CONFIG_SMALL("MPEG audio layer 2/3"),
> @@ -213,6 +240,7 @@ AVInputFormat ff_mp3_demuxer = {
>      .read_probe     = mp3_read_probe,
>      .read_header    = mp3_read_header,
>      .read_packet    = mp3_read_packet,
> +    .read_seek      = read_seek,
>      .flags          = AVFMT_GENERIC_INDEX,
>      .extensions     = "mp2,mp3,m2a", /* XXX: use probe */
>  };

I am no MP3 expert, though.

Regards,

-- 
  Nicolas George
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